The Internet can be a desirable alternative to those telephone users who wish to save on their telephone bills and can tolerate the occasional delays and dropouts or loss of quality of service due to data traffic congestion on the public Internet. Voice telephone calls over a network such as the Internet, referred to as Voice-over-IP ("VoIP"), allows callers to converse over the telephone with only limited use of the Public Switched Telephone Network ("PSTN") or General Switched Telephone Network ("GSTN") equipment provided by the local and long distance service providers. Rather than using the GSTN, VoIP calls are carried over the public Internet, thereby substantially avoiding the fees and charges levied by the long distance service providers who provide the GSTN equipment and service.
The GSTN establishes traditional circuit-switched connection between callers to continuously carry voice signals between the callers. A caller wishing to speak with another telephone subscriber picks up the telephone and dials the telephone number of the subscriber with which he is wishing to speak. According to the dialed telephone number, the GSTN establishes a circuit-switch connection using the telephony signaling and control protocols that have been established to setup dedicated circuit-switched connections over the hierarchy of switches and transmission equipment provided by the GSTN. The circuit-switched connection established by the GSTN is dedicated to one call that has exclusive access to the connection for the duration of the call.
In comparison to the dedicated circuit-switched connections established by the GSTN, computer networks such as the Internet provide voice communications, as well as multimedia communication such as text, graphics, video and audio, over a packet-based network. Rather than establishing a dedicated circuit-switch connection through the GSTN, a VoIP call establishes a virtual call connection between the two callers through the system of interconnected packet-based networks ("PBN") that make up the Internet, intranets and other digital networks that provide connectivity between users. The voice or multimedia information is broken up into packets that are transmitted over the different networks that carry the virtual connection.
In order to facilitate communication using the Internet, industry and international standards bodies have established sets of functional requirements, conventions or rules that govern the transmission of data over both telephone and packet switched computer networks. These functional requirements or rules are known in the art as "protocols." The implementation of protocols is necessary in order to bring order, and standardization, to the communications field and allow equipment of diverse manufacturers to be interoperable.
Some protocols are considered low level transmission media-related modulation protocols, such as modulation schemes implemented in a modem, for example V.34, V.22 bis, etc. Other protocols are considered higher level, and relate to such features as error control, transmission control protocols and network level routing and encapsulation of data. Examples of such protocols are the Point-to-Point Protocol (PPP), the Serial Line Interface Protocol (SLIP), and the Real-time Transport Protocol (RTP). The requirements of these latter protocols are typically prepared as a RFC "Request For Comment" document, circulated among and adopted by the industry. Sometimes other standards bodies such as the ITU eventually adopt the IETF standards as their standards as well. As an example, RTP (RFC-1889) has been placed into the ITU's H.225.0.
Developers have applied the various functions defined in protocols to develop devices and systems that improve the performance and capabilities of the Internet as well as of other types of data networks. One such device is a "gateway". Gateways allow dissimilar computer networks using different protocols and transmission rate capacities to interconnect by providing an interface that translates data between the different network formats. For example, one type of gateway is an Internet telephony gateway. An Internet telephony gateway is capable of receiving simultaneous incoming calls from the Public Switched Telephone Network and routing them to a data network. Internet telephony gateways may be used in VOIP systems, or Internet telephony systems, which permit virtual call connections for VOIP calls.
In VOIP calls, a first caller may place a telephone call using the caller's telephone or computer modem to a local Internet telephony gateway, which is connected to a PBN. The local gateway establishes one or more Internet sessions with a remote Internet telephony gateway. The remote gateway completes the virtual call connection by connecting to the second caller over a local telephone connection on the GSTN.
In order to communicate audio signals in an Internet-based telephone system, the gateway uses the audio signals received from the parties' telephones over the telephone network. These audio signals are typically pulse code modulated (PCM) signals according to the international G.711 standard. Audio signals coded in G.711 may need to be transcoded to G.723.1 or G.729 compressed audio signals to conserve bandwidth. The compressed audio signals are packetized and communicated in streams of packets over the Internet.
While there are cost benefits to be enjoyed by placing VOIP calls as opposed to traditional GSTN switched calls, callers may have to adjust to telephone connections that are different from the POTS connections to which they are accustomed. The virtual call connection provided by the Internet telephony system is different from the POTS connections because the gateways and the PBN replace a substantially continuous conductive path between the parties' telephones. In a voice telephone connection, POTS telephones on GSTN switched calls use the continuous conductive path to conduct audio signals from one telephone to another with almost no data processing other than the possible conversion of the analog voice signals to digital signals. In a VOIP virtual call connection, the gateways process the voice signals using a variety of protocols.
Because of the data processing of the voice signals used in Internet telephony systems, many features of the GSTN that relied on the ability to conduct signals along a substantially continuous path are lacking. One such feature is the communication of dual-tone multi-frequency/multiple frequency (DTMF/MF, hereinafter DTMF) tones between parties to a virtual call connection. U.S. Pat. No. 5,577,105 "TELEPHONE CALL ROUTING AND SWITCHING TECHNIQUES FOR DATA COMMUNICATIONS" to Baum et al., which is fully incorporated herein by reference, discloses the use of DTMFIMF signals for configuring calls by correlating the signals with communications, routing or applications protocols. In Baum et al., however, the DTMF/MF signals are not transported between two telecommunications devices that are connected over the PBN.
One reason why the transporting of DTMF/MF signals is difficult is that the G.723.1 and G.729 coding is based on a model of the human voice. Coding or decoding errors may occur because voice signals may contain frequencies similar to the frequencies of the DTMF tones. Encoding the DTMF tones and audio signals as packetized G.711 may reduce errors; however, packetized G.711 would make inefficient use of the network bandwidth.
It would be desirable to reliably transport DTMF signals over a wide-area network telephony system without the additional burden on the network.
One solution is to transmit DTMF signals in their own stream. The DTMF signals may be encoded as digits, which may then be packetized in a data stream and transported separately, or out-of-band, from the audio signal that contains both voice and DTMF signals. This solution has other advantages in that the DTMF signals may be integrated functionally into the telephony implementation as control signals that may, for example, permit data entry during call setup.
One problem with this solution is that during the processing of the DTMF detection, a 30-60 ms. skew will develop between the audio stream and the DTMF stream. Further, since the DTMF is sent as a separate stream to the remote gateway, this separate DTMF may experience greater delay than the audio stream while they are both being sent over the Internet. This is because the DTMF may be sent over a reliable transport protocol such as TCP, and the audio stream may be sent over an unreliable transport stream such as UDP. When the two streams (audio and DTMF) arrive at the remote gateway, they may have enough skew between them to be interpreted as two separate DTMF signals instead of the actual one DTMF signal. For this reason, the local Gateway will need to not only detect DTMF (in the audio stream) and regenerate the DTMF signal (into the separate DTMF stream) but also remove the DTMF stream from the original audio stream.
This additional processing of removing the DTMF from the audio stream requires that an additional 30-60 ms. delay be inserted between the point at which the audio stream is received from the PSTN at the Gateway and the point at which the audio stream is sent out to the Internet as packetized audio. This delay is necessary to permit detection and removal of DTMF signals from the audio signal. The problem with delaying the audio stream to detect and remove the DTMF from the audio stream is that a 30-60 ms. delay would result in a total round-trip delay of 60-20 ms. The human hearing can perceive a delay of about 300 ms. Because other processing will add additional fixed delay, it may not be acceptable to tolerate the 60-120 round trip delay.
Due to voicemail and services that use integrated voice response systems, telephony system should be able to reliably transport DTMF signals. It would be desirable for an Internet telephony system to provide the same capabilities as the POTS system. It would be particularly desirable to transport DTMF signals in a wide-area network telephony system without having to substantially delay the audio signal.